It might be premature as well as half baked - I am just testing out the
thought, however here is my thinking...
There is a folder called extensions in the dialplan folder. The nice
thing about this folder is that it is read in by a simple xml
preprocessing command that says go get *.xml
This was what we talked about on IRC a few days ago right?
/b
On Sep 21, 2008, at 2:45 AM, Christian Jensen wrote:
It might be premature as well as half baked - I am just testing out
the thought, however here is my thinking...
There is a folder called extensions in the dialplan folder.
The FREESWITCH version is the 1.0.1 release version. I have updated to
the snapshot from the 20th of September. But I get problems with the
./configure. The bootstrap.sh works OK after I have updated autoconf to
version 2.63 and automake to version 1.10.1. The error from ./configure
after half a
i just copied the dialplan given in wiki for record session but cant record
or lets say first i cant initiate the call only, sofia says codec not found
as soon as i hit dial on the dialer
can some1 help in the xml as im trying to record calls made from a specific
user whose account is created
Try automake 261
/b
On Sep 21, 2008, at 8:16 AM, Jon Bruel wrote:
The FREESWITCH version is the 1.0.1 release version. I have updated
to the snapshot from the 20th of September. But I get problems with
the ./configure. The bootstrap.sh works OK after I have updated
autoconf to version
The Aastra 55i should have STUN support, but not on the firmwares released
with the phone initially. Check your firmware version here:
http://www.aastratelecom.com/cps/rde/xchg/SID-3D8CCB6A-FE4A1E69/03/hs.xsl/21
669.htm
For the 480i, though, I think you're SOL.
- Darren
-Original
I have looked all over the web interface and can't find it. I have
the latest firmware too... weird. Still no rport support..?!?
/b
On Sep 21, 2008, at 11:45 AM, Darren Schreiber wrote:
The Aastra 55i should have STUN support, but not on the firmwares
released
with the phone initially.
Hi Everybody
I am building a conferencing service, and I was wondering if there is a
system variable that I can use, that holds the number of members in a
specific conference.
Is there a variable I can read (in java, or anywhere) that tells me this ?
If there is, is there a list of FreeSWITCH
Hi Brian, I have Aastra 55i, it does have STUN and Rport support, and I like
its rport support. On web interface, under network settings-Advanced
Network Settings-Rport (RFC 3581)Enabled for Rport
or STUN Server STUN Port
for STUN support, however I like rport better, which I can have call
you can see in the trace that the exchange is using alaw.
So the complaint about trans coding is because it's going from PCMU on leg a
to PCMA when talking to exchange.
please revert any changes, upgrade to current SVN trunk and try configuring
exchange to use PCMA
or confiigure FS to only use
Hi,
I need to create and destroy local_streams dynamically, that is, I need to
be changing the MOH of several fifo's in real-time, since local_stream
creates a thread that is broadcasting audio I would like to be able to
create and destroy them as I need them.
How can this be achieved?
This is not currently possible. It's something that could be added
but would require a rework of mod_local_stream
Mike
On Sep 21, 2008, at 3:15 PM, Cesar Cepeda wrote:
Hi,
I need to create and destroy local_streams dynamically, that is, I
need to be changing the MOH of several fifo’s in
ok i marked bypass media as well as proxy media to default which was like
commented, means marked as comment. With proxy media to enabled i used to
get the error of cant find codec but after making it all to default config,
im now getting this
2008-09-22 00:07:46 [ERR] mod_sndfile.c:175
On Sep 21, 2008, at 4:11 PM, xbipin wrote:
ok i marked bypass media as well as proxy media to default which was
like
commented, means marked as comment. With proxy media to enabled i
used to
get the error of cant find codec but after making it all to default
config,
im now getting
Thanks for an answer un Sunday ;)
Cesar Cepeda.
De: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] En nombre de Michael
Jerris
Enviado el: Sunday, September 21, 2008 2:59 PM
Para: freeswitch-users@lists.freeswitch.org
Asunto: Re: [Freeswitch-users] Creating and destroying local_stream
file doesnt exist but doesnt it need to create it by itself as u never know
whose gonna call when and where.
basically im looking for it to create the file and record in it so i can get
a different file for each user, whats the point creating the file and then
letting it record to it
Michael
On Sep 21, 2008, at 5:00 PM, xbipin wrote:
file doesnt exist but doesnt it need to create it by itself as u
never know
whose gonna call when and where.
basically im looking for it to create the file and record in it so i
can get
a different file for each user, whats the point creating
yes it does
Michael Jerris wrote:
On Sep 21, 2008, at 5:00 PM, xbipin wrote:
file doesnt exist but doesnt it need to create it by itself as u
never know
whose gonna call when and where.
basically im looking for it to create the file and record in it so i
can get
a different
That would def. solve the problem :P
/b
On Sep 21, 2008, at 2:03 PM, unknown wrote:
Hi Brian, I have Aastra 55i, it does have STUN and Rport support,
and I like its rport support. On web interface, under network
settings-Advanced Network Settings-Rport (RFC 3581)Enabled for Rport
or
STUN
just keep the same directory name and change the contents of the directory
as much as you want.
On Sun, Sep 21, 2008 at 3:34 PM, Cesar Cepeda [EMAIL PROTECTED] wrote:
Thanks for an answer un Sunday ;)
Cesar Cepeda.
*De:* [EMAIL PROTECTED] [mailto:
[EMAIL PROTECTED] *En nombre de
Hi,
I have a VERY simple FS setup.
1) Calls come in and go out via Vitelity. (I just route to their IP
address, don't even need a gateway.)
2) I have users registered remotely with anything from a single SIP
phone to an asterisk box.
3) Basically, I am a pass through of calls to and from
http://www.gbridge.com/ + FS = Awesome
-E
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I would transfer to 1000 XML default for example. This is why you
split inbound and outbound into two context's like you have it... but
since you're sending the call into default as 10 digits I suspect it
loops right back out.
Also can you please not hijack threads.
Click new message,
Yeppers.
On Sep 21, 2008, at 12:52 AM, Brian West [EMAIL PROTECTED] wrote:
This was what we talked about on IRC a few days ago right?
/b
On Sep 21, 2008, at 2:45 AM, Christian Jensen wrote:
It might be premature as well as half baked - I am just testing out
the thought, however here is
I have put the calltest.js in /usr/local/freeswitch/scripts and changed
sofia to openzap/default/[EMAIL PROTECTED] in the coding and i got the error:
Error:
2008-09-22 10:13:26 [ERR] switch_core_session.c:249
switch_core_session_outgoing_channel() Could not locate channel type openzap
2008-09-22
Hi Preetha,
you have use like this openzap/1/a/[EMAIL PROTECTED]
Thanks
Kris Anand
On Mon, Sep 22, 2008 at 10:19 AM, preetha Ayyappan
[EMAIL PROTECTED] wrote:
I have put the calltest.js in /usr/local/freeswitch/scripts and changed
sofia to openzap/default/[EMAIL PROTECTED] in the coding and
On Sep 22, 2008, at 12:49 AM, preetha Ayyappan wrote:
I have put the calltest.js in /usr/local/freeswitch/scripts and
changed sofia to openzap/default/[EMAIL PROTECTED] in the coding and
i got the error:
Error:
2008-09-22 10:13:26 [ERR] switch_core_session.c:249
thanks for your reply.it is working
--
Preetha.A
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